General governmental, as well as consumer, applications for remote audio collection require operation in many different, challenging environments such as indoor, outdoor, automobile and portable (body-carried or -worn), which experience a variety of bothersome conditions, such as wind, sand, dust, precipitation, radio frequency (RF) interference (e.g. from mobile communications), extreme temperatures, and acoustic noises. Limited scenarios have been addressed by prior devices, such as hands-free directional microphones for automobiles and small microphone arrays for computer workstations and hearing aids. Significant problems remain for prior devices to function effectively in the more general case. Prior devices also can not be easily scaled between small and large configurations for greater effectiveness, without significant impact on complexity, noise performance, power consumption, or architecture.
Directional microphones by definition selectively receive the sounds situated directly in-line with their (on-axis) look direction and have the ability to cancel or reject sounds coming from other (off-axis) directions. A microphone array can be used as a directional microphone system and consists of, in its simplest form, a plurality of microphones with appropriate processing of the audio signals from the microphones so as to accomplish the formation of a directional pick-up pattern. A traditional simple broadside microphone array is shown in FIG. 1.
Microphone arrays of this type, which use direct summation of the signals from the array of microphones, produce a directivity (i.e. width of the mainlobe of the pick-up pattern) which is dependent on the frequency. The directivity generally depends on the effective length of the array and the acoustic wavelength at the inspected frequency. Therefore, at low frequencies a lesser degree of directivity is achieved and the directivity increases with the frequency.
The lowest wavelength at which a microphone array can provide a certain degree of directivity is dependent on the overall length of the array. The highest frequency at which the pick-up pattern does not exhibit spatial aliasing (i.e. which causes loss of directional characteristics at high frequencies) depends on the distance between the microphones in the array.
Prior directional microphone array devices have been implemented in both analog and digital hardware circuitry, as well as in software (with appropriate audio capture hardware to collect and digitize the sound). Prior all-analog hardware implementations have been based on high-impedance resistive summation circuits. High-impedance summing circuits inherently have poor noise immunity, which in turn limits the maximum number of microphone elements that can be employed effectively as well as makes them very susceptible to electromagnetic interference.
Prior digital devices are generally complex, even for small systems such as those used in hearing aids. If many channels are digitized, then they quickly become impacted by the additional size (along with associated weight) and power requirements. These deficiencies result from the fact that in digital microphone array systems, each microphone channel must be digitized separately and synchronized with all other channels, carried to a central processor for beamforming and other filtering, and then reconverted to analog (sound) for the user to hear. Therefore, digital implementations suffer from a scaling problem in size, weight, power consumption, and cost as the array size increases to hundreds or thousands of microphones. As the number of channels increases, the amount of digital noise also increases. Additional shielding or other techniques are required, which further increase the weight and/or size. Furthermore, in traditional digital microphone array systems there is a central processor which performs the beamforming. For any given model digital processor, there is a maximum amount of data that can enter or exit the processor at any given time as well as a maximum number of instructions the processor can execute at any given time. Therefore, any given model digital processor has an inherent limit to how many audio channels it can accommodate.
The limit on the number of audio channels for a given model processor is a significant issue. For a microphone array to address most general cases, many microphone channels are required. As a practical matter, in challenging environmental scenarios with strong directional noise sources or diffuse noise, many microphone elements are needed if sidelobes of the pickup pattern are to be kept to a minimum while maintaining a narrow mainlobe (so as to reduce the pickup of off-axis interferences). This is the limitation inherent in microphone arrays used as hearing aids, for instance—all of the in-ear and on-ear implementations are inherently limited by how much weight, size, and heat can be accommodated by the wearer and therefore, even with sophisticated digital processing, digital hearing aid microphone arrays have limited directionality and sidelobe attenuation.
Prior devices have been constructed from electret or other types of microphones that have excellent sensitivities and do not require the type of phantom power used by studio microphones, but these microphones also have limited ranges of operation over temperature extremes, such as the military might encounter in hot deserts.
Microphone elements other than electrets have been tried in the prior art. Arnold, et al (JASA, 113, January 2003) published results of their construction of a low impedance, silicon (Micro-Electro-Mechanical Systems or MEMS based) element microphone array which mounted the microphones and centralized digital processor on a printed circuit board (PCB). The system by Arnold, et al, was novel in the sense that by using silicon microphone elements they lowered the overall system cost (as the individual microphones are cheaper) and did not have as many cables (because the audio signal was carried in the PCB traces). However, this array processing section was an all-digital implementation, so it did not take advantage of the microphones' analog electrical properties, nor did it provide for acoustic and vibrational damping, inter-microphone isolation, or wind protection. Being digital, it still inherently suffered from limitations in scalability regarding power consumption, complexity, heat, and weight. The scalability limits due to complexity are because of the inherent limits of any given model digital processor and its data bus. Extra stiffening components also had to be added to the printed circuit board to support its own weight.
Prior devices have incorporated single cables to transport microphone or beamformed audio within the system. Using multiplexing to implement a single cable audio transport within the system is well known in digital applications. However, employing digital multiplexing in large digital arrays is extremely difficult because of the complexity, cost, power requirements, distance limitations, and timing requirements of the digital circuitry. Digital multiplexing based implementations are not as versatile since each implementation must be designed for a specific (maximum) size array.
The single cable analog implementation described in the application by Soede et al (U.S. patent application Ser. No. 10/943,456, filed Sep. 17, 2004) employs a series of two-node summing networks each followed by a buffer. In their implementation, each stage of the series adds noise along with the signal of the additional microphone and therefore realizes no improvement in signal to noise ratio as more elements are added. This did reduce the power consumption and complexity of the device compared to all-digital implementations, but the solution created a scalability problem related to noise due to the use of the serial high impedance summing—the more stages, the more noise. This is not a concern for their targeted application, which was hearing aids, but becomes a limiting issue when scaling up to larger systems to address the general case.
The present inventor has in the past worked on a commercial product (conceived and implemented by the inventor along with other colleagues) that involved interconnecting multiple high impedance analog beamformers in a master/slave (hub/spoke) configuration. Although this particular implementation was significantly better than previous devices, it also suffered from noise susceptibility to RF interference, summing and other noises, as well as temperature restrictions (due to the electret microphones used).
Prior analog and digital devices have employed so called “aperture shading” to modify the pickup (beam) pattern. Rather than simply summing the microphone outputs (which might be individually time-delayed or not, depending on whether or not electronic steering is used), with aperture shading the signals are first multiplied by different gain factors (or weights) before summing. This extra pre-filtering step shades the aperture, allowing the designer to tradeoff beamwidth and sidelobe attenuation. This is analogous to choosing a window shape in 1-D (frequency) filter design. This is an effective technique, however it is a tradeoff and the designer must strike a balance between two desirable characteristics.
A highly directional audio system that can operate in a wide range of environments and be applied to various fixed, portable, and mobile applications needs to be physically and electrically robust, extremely power efficient, economical, inherently scalable, and noise immune while improving on directivity, sidelobe attenuation and audio quality. Previous implementations of analog and digital audio arrays have therefore not been able address all of these concerns simultaneously.
Several objects and advantages of the present invention are:
(a) to allow modular construction of a directional audio array that is highly scalable with no practical limit to the size (i.e. number of microphone elements) in the array;
(b) to provide significantly improved noise immunity compared to previous techniques;
(c) to allow low cost of construction, high reliability, high temperature operation, low overall system weight, and simplicity of operation;
(d) to significantly lower the effective internal system noise even when employing microphones with relatively high individual internal noise so as to receive, amplify, and reproduce low level on-axis sounds at intelligible levels;
(e) to allow integration of microphone elements, windscreen, protection from blowing sand and dust, moisture protection, vibration damping, acoustic shielding and isolation, structural rigidity and beamforming circuitry;
(f) to allow surface mounting of the sensors and all other electronic components for cost efficient automated fabrication;
(g) to allow use of a single cable and bus to serve as the interface to the entire array of modules and allow secondary parallel beamforming of additional modules;
(h) to provide additional off-axis noise rejection above and beyond standard beamforming;
(i) to modify the audio output in such a way as to allow the listener to more easily distinguish between off-axis and on-axis sounds by utilizing a psychoacoustic effect;
(j) to allow the creation of an easily scalable analog pre-beamformer on the front end of an analog or digital audio array system;
(k) to allow the efficient use of a sufficient number of sensors to simultaneously have high gain, high directivity, and high sidelobe attenuation;
(l) to allow the user to easily and immediately determine whether the sound source concerned is situated in the center of the main lobe, which is very important when using microphone arrays with a high degree of directivity;
(m) to allow uniform and non-uniform inter-microphone spacings so that the pick-up pattern can be manipulated by the designer;
(n) to allow the use of omni-directional or uni-directional microphone elements with appropriate mounting to make the pattern less or more directional; and
(o) to allow the extension of the low impedance summation and modular beamforming array techniques to include other audio transducers, such as ultrasonic and accelerometer as two examples.
Still further objects and advantages of this invention will become apparent from a consideration of the ensuing description and drawings.